SIP Registration Server NO IP or Port

1

Ok, what am I missing. I just want nodes to use a registration line to a server. All seam good, but I am not seeing nodes IP address or Port. The following is the code.

Remote Node registration line: register => 1800:12345@47.187.xxx.xxx

-=-=-=-=-=-=-=-

; SIP Configuration for Asterisk

[general]
context = default ; Default context for incoming calls
allowguest=no
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 127.0.0.1 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = no ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
[1800]
type=peer
host=dynamic
sercret=123456
username=KG5RDF
disallow=all
allow=ulaw
transfer=no

-=-=-=-=-=-=-=-

Output in CLI:

Name/username Host Dyn Nat ACL Port Status
1800/KG5RDF (Unspecified) D 0 Unmonitored
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 1 offline]

2

Changed peer to friend to see user info. Fixed secret spelling too. Still no ip or port number.

3
  • Name : 1800
    Secret :
    MD5Secret :
    Context : default
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox :
    VM Extension : asterisk
    LastMsgsSent : 32767/65535
    Call limit : 0
    Dynamic : Yes
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : RFC3581
    ACL : No
    T38 pt UDPTL : No
    CanReinvite : Yes
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : No
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : (Unspecified) Port 0
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: KG5RDF
    SIP Options : (none)
    Codecs : 0x4 (ulaw)
    Codec Order : (ulaw:20)
    Auto-Framing: No
    Status : Unmonitored
    Useragent :
    Reg. Contact :
4

‘NODES’ do not use ‘SIP’ format. They use IAX.

Can you say better what you are trying to do exactly ?

5

I am needed a registration server that will take the stand node registration line:

register => 1800:12345@×××.×××.×××.×××

I noticed sip does this. It shows 1800 as a peer, but does not show 1800 ip address or port. I was going to parse the file to update the database extnodes.

6

Yes SIP does this for SIP devices.
IAX does this for IAX devices.
What is the device ?
Nodes are IAX
SIP devices are not going to communicate with IAX devices. Different protocol.

You can register nodes in rpt.conf file under [NODES]

7

I was looking for only inbound sip from remote node registration line. I gotten a node to show its connected as peer using remote node registration line, but no ip or port shown. It was a thought to try. Do not want to use [nodes] in rpt.conf. I need a real time registration server. I need to update the database as ip’s change and if they are connected to the system. It about 100 nodes. I manually created download files, for the server to pull and update astdb and extnodes. I need to automate this process. I rewritten the server code and system is up and running. Just need to do the registration server.

8

I se a little better now.
You might call that a peer trunk or peer connection depending on methods…
The terminology you used confused me.

You are attempting to trunk/peer some other server to this one.

From an asterisk cli>
sip show peers

9

Yes, three servers. I did get the registration server working and seeing ip’s, callsigns, and ports via iax. Will revisit the sip later today for servers.